A few things.
1. Make it work in alsa, do not mix it with PA technology. The combine-sink module uses adaptive resampling to merge two soundcards. I am not sure it is what you want - it will very likely introduce some (random?) delay between the two card streams.
2. Do not complicate things by using the plugin names for your devices. This may or may not work, I do not know. The best is to avoid such situation. Call your devices differently - e.g. multidev, ttabledev, etc.
3. How did you check there was no virtual soundcard created? Run
which lists hardware as well as virtual devices.
Configuring PA with that virtual device should be:
in ~/.config/pulse/default.pa or the global default.pa file - Jürgen knows most about configuring PA.
Is the ttabledev actually needed when all the channel mappings are 1->1?
1. Make it work in alsa, do not mix it with PA technology. The combine-sink module uses adaptive resampling to merge two soundcards. I am not sure it is what you want - it will very likely introduce some (random?) delay between the two card streams.
2. Do not complicate things by using the plugin names for your devices. This may or may not work, I do not know. The best is to avoid such situation. Call your devices differently - e.g. multidev, ttabledev, etc.
3. How did you check there was no virtual soundcard created? Run
Code:
aplay -Lwhich lists hardware as well as virtual devices.
Configuring PA with that virtual device should be:
Code:
load-module module-alsa-sink device=multidevin ~/.config/pulse/default.pa or the global default.pa file - Jürgen knows most about configuring PA.
Is the ttabledev actually needed when all the channel mappings are 1->1?
In answer to my question whether the pacmd command persists after a reboot, no.
I don't understand. The point of this is to get two sound cards to work as one in PAX Rack, so why do you say, "make it work in ALSA"?
I ran aplay -L and got more than I expected;
This is frustrating as hell. I have tried every variant found in the ALSA project web pages, changing things and testing but with no success.
.
I don't understand. The point of this is to get two sound cards to work as one in PAX Rack, so why do you say, "make it work in ALSA"?
I named the devices 101 and 202 in the udev ALSA rules file and that worked. I tried to create a virtual sound card named 303 but it would not show in PAX Rack or PulseAudio Volume Control (pavu).
I ran aplay -L and got more than I expected;
How do I get rid of all of the detritus?
I don't know but when I get rid of all of the excess, I will use
to find out.
This is frustrating as hell. I have tried every variant found in the ALSA project web pages, changing things and testing but with no success.
Alsa is a layer beneath PA. PA itself cannot merge soundcards (as already discussed with Jürgen). Merge the soundcards in alsa and tell PA to use this PCM device.
By default PA offers only devices, not PCMs. Look at the difference between parameters -l and -L in man aplay.
The output is correct. All these PCM (virtual) devices are defined for each of your cards in default alsa configuration in /usr/share/alsa/pcm.
But I do not see the PCM you defined in your .asoundrc. Is your config correct? Place it here please.
No need to do so, just do not use the other PCMs.
The reason is you do not know how the stuff works internally, what is the relation between PA and alsa, therefore you do not know what to configure and what to look for. No reason to be frustrated, you are learning and you will do fine.
One more important thing - test the newly created 10ch PCM device in alsa before re-configuring PA to use it.
For details see man speaker-test.
The device must work OK in alsa first, only then can it be used by upper layers (PA). That is the standard troubleshooting procedure - testing/putting into operation changes layer by layer. Your final sound chain will have quite a few layers where any problem will defunct the whole chain
Playback app -> PA input -> PAX filters -> PA output -> alsa-lib virtual PCMs -> alsa raw (hw:X) devices -> alsa kernel modules/drivers -> HW soundcards.
Code:
speaker-test -c 10 -D multidevFor details see man speaker-test.
The device must work OK in alsa first, only then can it be used by upper layers (PA). That is the standard troubleshooting procedure - testing/putting into operation changes layer by layer. Your final sound chain will have quite a few layers where any problem will defunct the whole chain
Playback app -> PA input -> PAX filters -> PA output -> alsa-lib virtual PCMs -> alsa raw (hw:X) devices -> alsa kernel modules/drivers -> HW soundcards.
Can we go back a step to see if I understand this?
In the udev file, the sound cards were given the names of "101" and "202". Is this correct?
Running cat /proc/asound/cards gives this;
In the udev file, the sound cards were given the names of "101" and "202". Is this correct?
Running cat /proc/asound/cards gives this;
I don't understand this. There isn't a 10 channel PCM device in ALSA, at least I can't see it.
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Your cards list shows the udev rule works correctly. This is the information that command aplay -l would provide - listing cards.
Right, the devices "multidev" and "ttabledev" you configured in .asoundrc are 12 channel (6 + 6), my fault. These are not cards, but PCM devices defined on top of the cards. Raw PCM devices for your cards are called "hw:101", "hw:102", "hw:HDMI", "hw
CH". These are the device strings you would use in alsa application configs (e.g. in PA output device config).Speaker test;
frank@MintMusic:~$ speaker-test -c 10 -D multidev
speaker-test 1.1.3
Playback device is multidev
Stream parameters are 48000Hz, S16_LE, 10 channels
Using 16 octaves of pink noise
ALSA lib pcm.c:2495
Playback open error: -2,No such file or directory
frank@MintMusic:~$ speaker-test -c 10 -D combcm106
speaker-test 1.1.3
Playback device is combcm106
Stream parameters are 48000Hz, S16_LE, 10 channels
Using 16 octaves of pink noise
ALSA lib pcm_multi.c:1225
Playback open error: -22,Invalid argument
frank@MintMusic:~$ speaker-test -c 12 -D multidev
speaker-test 1.1.3
Playback device is multidev
Stream parameters are 48000Hz, S16_LE, 12 channels
Using 16 octaves of pink noise
ALSA lib pcm.c:2495
Playback open error: -2,No such file or directory
frank@MintMusic:~$
Why did you mess up so much the original .asoundrc in https://www.diyaudio.com/forums/pc-...-crossovers-using-pax-rack-2.html#post5785120 which looked OK, except for the device names, e.g. multi -> multidev, ttable -> ttabledev?
Your last version does not make any sense.
Your last version does not make any sense.
Yes and I removed the full stops (periods) and it still did not result in a virtual card being available in PAX Rack.
As I said above, I went through that version and Google searched other methods, and tried them all with as many variations as I could imagine to try to get something to show up as a virtual card. Result; ZILCH.
The idea of a single box to play music and act as a crossover with remote implementation is very attractive but is looking like a step too far.
Is there a website or program that will parse this type of Linux code and detect syntax errors?
As I said above, I went through that version and Google searched other methods, and tried them all with as many variations as I could imagine to try to get something to show up as a virtual card. Result; ZILCH.
The idea of a single box to play music and act as a crossover with remote implementation is very attractive but is looking like a step too far.
Is there a website or program that will parse this type of Linux code and detect syntax errors?
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As I said, work step by step. First the speaker-test for the merged PCM device must pass. I.e. your .asoundrc must be correct. Only then can you move one layer up - pulseaudio.
Rename the devices and fix the periods in the .asoundrc in the post I linked, and post it here along with speaker-test results testing both the devices configured in your .asoundrc.
Speaker-test gives you some info about the problem, PA will just avoid your device alltogether - no troubleshooting info you need.
Rename the devices and fix the periods in the .asoundrc in the post I linked, and post it here along with speaker-test results testing both the devices configured in your .asoundrc.
Speaker-test gives you some info about the problem, PA will just avoid your device alltogether - no troubleshooting info you need.
The results for the speaker test are above, though I think I did not code it properly.
BUT, the real but, I do not understand what this means;
linux - pulseaudio - simultanious - add two devices - Unix & Linux Stack Exchange
BUT, the real but, I do not understand what this means;
I know nothing about Linux coding at all, so the quoted text means little, if anything, to me. I was hoping to take some sample code, rejig it and see if it works. I can't see any light at the end of the tunnel at this stage, despite having spent days on it.
linux - pulseaudio - simultanious - add two devices - Unix & Linux Stack Exchange
You sent speaker-test for a totally corrupted .asoundrc which does not make any sense.
My last attempt:
1) Rename your card in udev rules from numerical 101, 202 to non-numerical card101, card202 . If you use numerical names, alsa does not know if you mean a card name or card ID (your card named 101 has ID 0 in https://www.diyaudio.com/forums/pc-...-crossovers-using-pax-rack-3.html#post5798661 )
Check with aplay -l the cards have correct names card101 and card202 (in square brackets)
2) Use the following .asoundrc:
3. Post output of
My last attempt:
1) Rename your card in udev rules from numerical 101, 202 to non-numerical card101, card202 . If you use numerical names, alsa does not know if you mean a card name or card ID (your card named 101 has ID 0 in https://www.diyaudio.com/forums/pc-...-crossovers-using-pax-rack-3.html#post5798661 )
Check with aplay -l the cards have correct names card101 and card202 (in square brackets)
2) Use the following .asoundrc:
Code:
pcm.multidev {
type multi;
slaves.a.pcm "hw:card101";
slaves.a.channels 6;
slaves.b.pcm "hw:card202";
slaves.b.channels 6;
bindings.0.slave a;
bindings.0.channel 0;
bindings.1.slave a;
bindings.1.channel 1;
bindings.2.slave a;
bindings.2.channel 2;
bindings.3.slave a;
bindings.3.channel 3;
bindings.4.slave a;
bindings.4.channel 4;
bindings.5.slave a;
bindings.5.channel 5;
bindings.6.slave b;
bindings.6.channel 0;
bindings.7.slave b;
bindings.7.channel 1;
bindings.8.slave b;
bindings.8.channel 2;
bindings.9.slave b;
bindings.9.channel 3;
bindings.10.slave b;
bindings.10.channel 4;
bindings.11.slave b;
bindings.11.channel 5;
}
pcm.ttabledev {
type route;
slave.pcm "multidev";
ttable.0.0 1;
ttable.1.1 1;
ttable.2.2 1;
ttable.3.3 1;
ttable.4.4 1;
ttable.5.5 1;
ttable.6.6 1;
ttable.7.7 1;
ttable.8.8 1;
ttable.9.9 1;
ttable.10.10 1;
ttable.11.11 1;
}3. Post output of
Code:
speaker-test -c 12 -D multidevVery good, your 12ch alsa PCM device looks working OK. You should see the pink noise coming from each channel as it is being played by speaker-test.
Now you have to tell PA to use it. This is really Jürgen's territory, just two commands:
Adding the new PCM device to the PA list of devices offered/available:
Switching PA output to this device:
IMO then PAXOR should see the 12 channels of the output device, but for that I suggest to move to the main PAXOR thread and talk to Jürgen (TFive).
Now you have to tell PA to use it. This is really Jürgen's territory, just two commands:
Adding the new PCM device to the PA list of devices offered/available:
Code:
pactl load-module module-alsa-sink device="multidev"Switching PA output to this device:
Code:
pacmd set-default-sink alsa_output.multidevIMO then PAXOR should see the 12 channels of the output device, but for that I suggest to move to the main PAXOR thread and talk to Jürgen (TFive).
Why not use alsa_in, alsa_out
Ubuntu Manpage:
alsa_in, alsa_out - Jack clients that perform I/O with an alternate audio interface
I tried it with 4 different sound devices at the same time on a HTPC with KXStudio
The sound comes out for all these devises but not 100% synchron.
But with some work and a delay-(VST)-plugin this problem can be solved.
Regards
Ubuntu Manpage:
alsa_in, alsa_out - Jack clients that perform I/O with an alternate audio interface
I tried it with 4 different sound devices at the same time on a HTPC with KXStudio
The sound comes out for all these devises but not 100% synchron.
But with some work and a delay-(VST)-plugin this problem can be solved.
Regards
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